• Journal of the Institute of Convergence Signal Processing
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• v.11 no.3
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• pp.238-243
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• 2010

A new implementation method for MPDSAP(Maximally Polyphase Decomposed Subband Affine Projection) adaptive filter is proposed. The affine projection(AP) adaptive filter achieves fast convergence speed, however, its implementation is so expensive because of the matrix inversion for a weight-updating of adaptive filter. The maximally polyphase decomposed subband filtering allows the AP adaptive filter to avoid the matrix inversion, moreover, by using a pipelining technique, the simple subband structured AP is suitable for VLSI implementations concerning throughput, power dissipation and area. Computer simulations are presented to verify the performance of the proposed algorithm.

• Journal of Biomedical Engineering Research
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• v.8 no.1
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• pp.1-6
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• 1987

In this study, we described the result of computer simulation which is real time extraction of FECG and FHR using adaptive digital signal processing. And we designed 2 channel cassette interface circuit to save FECG signal.

• ETRI Journal
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• v.34 no.5
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• pp.655-665
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• 2012

Based on wideband precoding (WBP) in the multiple-input multiple-output orthogonal frequency division multiplexing system, an adaptive nonuniform codebook is presented in this paper. The relationship between the precoder distribution and spatial correlation is analyzed at first. A closed-form expression based on overlapped isosceles triangles is proposed as an approximation of the precoder distribution. Then, the adaptive codebook design is derived with the approximate distribution to minimize quantization errors. The capacity and bit error rate performance demonstrate that the adaptive codebook with WBP outperforms the conventional fixed uniform codebook.

• Journal of KSNVE
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• v.8 no.4
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• pp.749-756
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• 1998

기어에서 충격성 진동 및 소음은 치차의 이상과 연관이 있다. 따라서 충격 진동 및 소리는 기어의 이상 진단에 사용되어 질 수 있다. 또한 이들 충격파를 조기에 정확하게 탐지하여 기어의 이상을 진단하면 완전 파손을 방지할 수 있다. 그러나 주변 소음 및 노이즈 신호 때문에 객관적이 충격파의 탐지가 어렵기 때문에, 본 논문은 이러한 숨겨진 충격 신호를 능동 신호 처리 기법을 이용하여 조기에 찾아내고 이것을 시간-주파수 영역에서 해석하였다.

• 제어로봇시스템학회:학술대회논문집
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• 1990.10a
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• pp.580-585
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• 1990

This paper presents a real time FECG signal monitoring system in which an adaptive multichannel noise canceller is implemented using a Texas Instruments TMS32020 digital signal processor. Abdominal ECG signal is applied as the desired output and 3 chest ECG signals as the reference input signals of the adaptive multichannel noise canceller whose coefficients are updated using the LMS algorithms.

• Journal of Advanced Navigation Technology
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• v.11 no.3
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• pp.319-328
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• 2007

In this paper we implemented the Real Time Contrast Enhancer for image quality enhancement of moving picture. Also we proposed adaptive contrast control method that use mean and variance of input video signal. The Designed the contrast Enhancer is measured in comparison with conventional pictures and interfaced to 30inch TFT LCD TV.

• Journal of electromagnetic engineering and science
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• v.8 no.2
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• pp.52-58
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• 2008

This paper concerns with signal cancellation problem in a linearly constrained adaptive array processor in coherent environment. Alternate mainbeam nulling approach was proposed to prevent the signal cancellation phenomenon. The linearly constrained LMS algorithm with a unit gain constraint and that with a null constraint in the direction of the desired signal is alternately implemented to reduce the signal interaction between the desired signal and the interferences, which is the main cause of the signal cancellation. It is shown that the proposed method performs better than a conventional method.

• The Transactions of the Korean Institute of Electrical Engineers P
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• v.64 no.4
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• pp.267-272
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• 2015

Active noise control (ANC) is a method of cancelling a noise signal in an acoustic cavity by generating an appropriate anti-noise signal via canceling loudspeakers. The continuous progress of ANC involves the development of improved adaptive signal processing algorithms, transducers, and DSP hardware. In this paper, the convergence behavior and the stability of the FxLMS algorithm in ANC systems with real-time implementation is proposed. Specially, The advanced DSP H/W with dual core(DSP+ARM) and API(application programming interface) S/W programming was developed to improve the real-time implementation performance under the FxLMS algorithms of input noise such as road noise environment. The experimental results are found to be in good agreement with the theoretical predictions.

• Speech Sciences
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• v.8 no.2
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• pp.73-81
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• 2001

In this paper, we focus on the real time implementation of the single channel adaptive noise canceller(ANC) by using TMS320C30 EVM board. The implemented single channel adaptive noise canceller is based on a reference paper [1] in which it is simulated by using the recursive average magnitude difference function(AMDF) to get a properly delayed input speech on a sample basis as a reference signal and normalized least mean square(NLMS) algorithm. To certify results of the real time implementation, we measured the processing time of the ANC and enhancement ratio according to various signalto-noise ratios(SNRs). Experimental results demonstrate that the processing time of the speech signal of 32ms length with delay estimation of every 10 samples is about 26.3 ms, and almost the same performance as given in [1] is obtained with the implemented system.

• The Journal of Information Technology
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• v.2 no.2
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• pp.127-138
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• 1999

The wavelet transform is widely used in signal processing application. In this paper, a wavelet domain adaptive algorithm(WPTNLMS) is derived and its performances are evaluated in non-stationary environment. Where the input signals are decomposed by the wavelet packet transform for the multi-resolution adaptive processing. And the NLMS is used as an adaptive algorithm in wavelet domain. The proposed technique is applied to noise cancellation of the Doppler signal which is added with white Gaussian noise.